Traditionally, telephony communications within the United States were handled by the public switched telecommunications network (PSTN). The PSTN can be characterized as a network designed for voice communications, primarily on a circuit-switched basis, with full interconnection among individual networks. The PSTN network is largely analog at the local loop level, digital at the backbone level, and generally provisioned on a wireline, rather than a wireless, basis. The PSTN includes switches that route communications between end users. Circuit switches are the devices that establish connectivity between circuits through an internal switching matrix. Circuit switches set connections between circuits through the establishment of a talk path or transmission path. The connection and the associated bandwidth are provided temporarily, continuously, and exclusively for the duration of the session, or call. While developed to support voice communications, circuit switches can support any form of information transfer (e.g., data and video communications).
In a traditional PSTN environment, circuit switches include central office (CO) exchanges, tandem exchanges, access tandem exchanges, and international gateway facilities. Central offices, also known as exchanges, provide local access services to end users via local loop connections within a relatively small area of geography known as an exchange area. In other words, the CO provides the ability for a subscriber within that neighborhood to connect to another subscriber within that neighborhood. Central offices, also known as end offices, reside at the terminal ends of the network. In other words, COs are the first point of entry into the PSTN and the last point of exit. They are also known as class 5 offices, the lowest class in the switching hierarchy. A class 5 telephone switch communicates with an analog telephone using the analog telephony signals in the well-known analog format. The class 5 telephone switch provides power to the telephone; detects off-hook status of the telephone and provides a dial tone in response; detects dual-tone multi-frequency signals from the caller and initiates a call in the network; plays a ringback tone to the caller when the far-end telephone is ringing; plays a busy tone to the caller when the far-end telephone is busy; provides ring current to the telephone on incoming calls; and provides traditional telephone services such as call waiting, call forwarding, caller ID, etc.
Referring to FIG. 1, a first local exchange (i.e., central office) 100 and a second local exchange 200 are connected by PSTN network 150. When placing a telephone call from the first local exchange 100 (i.e., the originating end of the network connection) to the second local exchange (i.e., the receiving end of the network connection), analog signals generated by telephone 115 are converted by local exchange 100 into digital signals for transmission over the digital backbone of the PSTN network 150. Likewise, digital signals received over the digital backbone of the PSTN network 150 are converted by local exchange 200 into analog signal for transmission to telephone 215. The conversion of analog signals to digital signals (as depicted in boxes 112 and 212) is also referred to as coding, and the conversion of digital signals to analog signals (as depicted in boxes 113 and 213) is referred to as decoding. Coding and decoding are also referred to as sampling. The equipment used to convert analog to digital (and vice-versa) is commonly referred to as a CODEC, identified by reference numeral 114 and 214. In a PSTN, CODECs typically reside on line cards within the central office. CODECs 114 and 214 convert analog signals to fixed-rate digital samples, and thus a sampling clock (identified by reference numerals 116 and 216) is required at both the transmitting and receiving end of the network (i.e., the network endpoints) in order to synchronize the coding and decoding of the digital data stream by the CODECs. Once a network connection is established, both the end of the network from which the call is originating (i.e., the originating end) and the end of the network which receives the call (i.e., the target end) transmit and receive data across the network in order for a contemporaneous conversation to occur. Thus, it is important to synchronize the sampling clocks (which control the sampling rate of the CODECs) so that an exact number of samples are transmitted and received by the network endpoints over a given time interval. If the sampling clocks are not synchronized, a frequency offset (i.e., error) will occur between the network endpoints, resulting in overflows or underflows (i.e., timing slips) of samples at the network endpoints. These timing slips reduce the quality of voice transmission and are particularly disruptive and troublesome to data transmission, where error tolerance is extremely tight. In a PSTN network, timing slips are prevented by synchronizing sampling clocks 116 and 216 to an embedded reference sample clock (represented by reference numeral 117) that is extractable from the first layer physical interface of the PSTN, which is typically a T1 or SONET OC3 fiber optic interface.
In an effort to increase the amount and speed of information transmitted across networks, the telecommunications industry is shifting toward broadband packet networks which are designed to carry a variety of services such as voice, data, and video. For example, asynchronous transfer mode (ATM) networks have been developed to provide broadband transport and switching capability between local area networks (LANs) and wide area networks (WANs). The Sprint ION network is a broadband network that is capable of delivering a variety of services such as voice, data, and video to an end user at a residential or business location. The Sprint ION network has a wide area IP/ATM or ATM backbone that is connected to a plurality of local loops via multiplexors. Each local loop carries ATM over ADSL (asymmetric digital subscriber line) traffic to an integrated service hub (ISH), which may be at either a residential or a business location.
An ISH is a hardware component that links business or residential user devices such as telephones and computers to the broadband, wide area network through a plurality of user interfaces and at least one network interface. A suitable ISH is described in U.S. Pat. No. 6,272,553 entitled “Multi-Services Communications Device,” issued on Aug. 7, 2001, which is incorporated by reference herein in its entirety. The network interface typically is a broadband network interface such as ADSL, T1, or HDSL-2. Examples of user interfaces include telephone interfaces such as plain old telephone system (POTS) ports (also referred to as jacks) for connecting telephones, fax machines, modems, and the like to the ISH; computer interfaces such as ethernet ports for connecting computers and local area networks to the ISH; and video ports such as RCA jacks for connecting video players, recorders, monitors, and the like to the ISH.
In providing telephony services over a broadband network, the ISH communicates with a service manager. This connection between the ISH and the network element is typically an ATM connection, which is much different from the traditional analog line to the local switch. ATM connections usually do not support analog telephony signals, such as off-hook, dial tone, and busy signals. Therefore, the ISH must provide many of the telephony functions traditionally provided by the telephone provider central office such as detecting off-hook and on-hook connections as well as providing the telephones with dial tone, ring voltage (sometimes referred to as ring current), ringback, and busy signals. The terms off-hook and off-hook condition as used herein are generic terms meaning that a user device (whether telephone, facsimile machine, modem, etc.) connected to a telephone line is attempting to access and use the line.
Another example of such a central office function being provided by the ISH is the coding and decoding of analog and digital signals. As shown in FIG. 2, ATM network 175 comprises a first endpoint 300 and a second endpoint 400. In an ATM network, endpoints 300 and 400 typically each comprise an ISH (reference numerals 310 and 410) connecting telephones 315 and 415 to ATM network 175. The coding and decoding functions are provided by CODECs 314 and 414 residing within integrated services hubs 310 and 410, respectively. Integrated services hubs 310 and 410 each contain a sampling clock (reference numerals 316 and 416, respectively) that controls the rate of sampling by the CODEC. Sampling clocks 316 and 416, residing at the endpoints of the network connection, need to be synchronized in order to prevent timing slips. Unfortunately, many newer ATM and other broadband packet networks do not currently support an embedded reference sample clock, and thus there is no common reference clock for the endpoint sampling clocks to extract and synchronize upon.
As shown in FIG. 2, one solution is to use local sampling clocks 316 and 416 that are close but not exactly equal at each endpoint. The drawback to this solution is that since the local sampling clocks are not synchronized, frequency offset or errors will occur between endpoints, resulting in reduced quality from underflows or overflows (i.e., timing slips) of samples at both endpoints. Using highly accurate clocks at the endpoints can minimize the frequency offset, but this is typically cost prohibitive, especially in an ISH that must be distributed to a large number of customers.
A variation of this solution, as shown in FIG. 3, involves extracting from the network connection a reference sampling rate representing the rate of sampling occurring at the end of the network connection opposite from the end connected to the integrated services hub and adjusting the sampling rate in the integrated services hub to about equal the reference sampling rate. The reference sampling rate may be an embedded signal, but preferably is extrapolated from the arrival rate of incoming cells to the integrated services hub. Extrapolation is achieved by monitoring the fill level of incoming cells received into an incoming cell buffer, increasing the sampling rate in the integrated service hub in response to an increase in the fill level of the incoming cell buffer above the midpoint, and decreasing the sampling rate in the integrated services hub in response to a decrease in the fill level of the incoming cell buffer below the midpoint.